gecko/content/media/webrtc/MediaEngineWebRTCAudio.cpp

225 lines
4.7 KiB
C++

/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "MediaEngineWebRTC.h"
#define CHANNELS 1
#define ENCODING "L16"
#define DEFAULT_PORT 5555
#define SAMPLE_RATE 256000
#define SAMPLE_FREQUENCY 16000
#define SAMPLE_LENGTH ((SAMPLE_FREQUENCY*10)/1000)
namespace mozilla {
/**
* Webrtc audio source.
*/
NS_IMPL_THREADSAFE_ISUPPORTS0(MediaEngineWebRTCAudioSource)
void
MediaEngineWebRTCAudioSource::GetName(nsAString& aName)
{
if (mInitDone) {
aName.Assign(mDeviceName);
}
return;
}
void
MediaEngineWebRTCAudioSource::GetUUID(nsAString& aUUID)
{
if (mInitDone) {
aUUID.Assign(mDeviceUUID);
}
return;
}
nsresult
MediaEngineWebRTCAudioSource::Allocate()
{
if (mState != kReleased) {
return NS_ERROR_FAILURE;
}
mVoEBase->Init();
mVoERender = webrtc::VoEExternalMedia::GetInterface(mVoiceEngine);
if (!mVoERender) {
return NS_ERROR_FAILURE;
}
mChannel = mVoEBase->CreateChannel();
if (mChannel < 0) {
return NS_ERROR_FAILURE;
}
// Check for availability.
webrtc::VoEHardware* ptrVoEHw = webrtc::VoEHardware::GetInterface(mVoiceEngine);
if (ptrVoEHw->SetRecordingDevice(mCapIndex)) {
return NS_ERROR_FAILURE;
}
bool avail = false;
ptrVoEHw->GetRecordingDeviceStatus(avail);
if (!avail) {
return NS_ERROR_FAILURE;
}
// Set "codec" to PCM, 32kHz on 1 channel
webrtc::VoECodec* ptrVoECodec;
webrtc::CodecInst codec;
ptrVoECodec = webrtc::VoECodec::GetInterface(mVoiceEngine);
if (!ptrVoECodec) {
return NS_ERROR_FAILURE;
}
strcpy(codec.plname, ENCODING);
codec.channels = CHANNELS;
codec.rate = SAMPLE_RATE;
codec.plfreq = SAMPLE_FREQUENCY;
codec.pacsize = SAMPLE_LENGTH;
codec.pltype = 0; // Default payload type
if (ptrVoECodec->SetSendCodec(mChannel, codec)) {
return NS_ERROR_FAILURE;
}
// Audio doesn't play through unless we set a receiver and destination, so
// we setup a dummy local destination, and do a loopback.
mVoEBase->SetLocalReceiver(mChannel, DEFAULT_PORT);
mVoEBase->SetSendDestination(mChannel, DEFAULT_PORT, "127.0.0.1");
mState = kAllocated;
return NS_OK;
}
nsresult
MediaEngineWebRTCAudioSource::Deallocate()
{
if (mState != kStopped && mState != kAllocated) {
return NS_ERROR_FAILURE;
}
mVoEBase->Terminate();
mVoERender->Release();
mState = kReleased;
return NS_OK;
}
nsresult
MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
{
if (!mInitDone || mState != kAllocated) {
return NS_ERROR_FAILURE;
}
if (!aStream) {
return NS_ERROR_FAILURE;
}
mSource = aStream;
AudioSegment* segment = new AudioSegment();
segment->Init(CHANNELS);
mSource->AddTrack(aID, SAMPLE_FREQUENCY, 0, segment);
mSource->AdvanceKnownTracksTime(STREAM_TIME_MAX);
mTrackID = aID;
if (mVoEBase->StartReceive(mChannel)) {
return NS_ERROR_FAILURE;
}
if (mVoEBase->StartSend(mChannel)) {
return NS_ERROR_FAILURE;
}
// Attach external media processor, so this::Process will be called.
mVoERender->RegisterExternalMediaProcessing(mChannel, webrtc::kRecordingPerChannel, *this);
mState = kStarted;
return NS_OK;
}
nsresult
MediaEngineWebRTCAudioSource::Stop()
{
if (mState != kStarted) {
return NS_ERROR_FAILURE;
}
if (!mVoEBase) {
return NS_ERROR_FAILURE;
}
mVoERender->DeRegisterExternalMediaProcessing(mChannel, webrtc::kRecordingPerChannel);
if (mVoEBase->StopSend(mChannel)) {
return NS_ERROR_FAILURE;
}
if (mVoEBase->StopReceive(mChannel)) {
return NS_ERROR_FAILURE;
}
mState = kStopped;
return NS_OK;
}
nsresult
MediaEngineWebRTCAudioSource::Snapshot(PRUint32 aDuration, nsIDOMFile** aFile)
{
return NS_ERROR_NOT_IMPLEMENTED;
}
void
MediaEngineWebRTCAudioSource::Shutdown()
{
if (!mInitDone) {
return;
}
if (mState == kStarted) {
Stop();
}
if (mState == kAllocated) {
Deallocate();
}
mVoEBase->Release();
mState = kReleased;
mInitDone = false;
}
typedef WebRtc_Word16 sample;
void
MediaEngineWebRTCAudioSource::Process(const int channel,
const webrtc::ProcessingTypes type, sample* audio10ms,
const int length, const int samplingFreq, const bool isStereo)
{
ReentrantMonitorAutoEnter enter(mMonitor);
nsRefPtr<SharedBuffer> buffer = SharedBuffer::Create(length * sizeof(sample));
sample* dest = static_cast<sample*>(buffer->Data());
for (int i = 0; i < length; i++) {
dest[i] = audio10ms[i];
}
AudioSegment segment;
segment.Init(CHANNELS);
segment.AppendFrames(
buffer.forget(), length, 0, length, nsAudioStream::FORMAT_S16_LE
);
mSource->AppendToTrack(mTrackID, &segment);
return;
}
}